On Sun, Oct 19, 2008 at 4:11 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > First, I think is better to to have SIP/vitel-outbound/${EXTEN} than > having SIP/[EMAIL PROTECTED] > And try issuing SIP SET DEBUG on the cli to see what happens when making the > call, post back what you see making calls from 101 to 102 and 102 to 101. > Having the sip.conf sould help on getting whats going on.
Here are the relevant parts of the sip.conf [general] register => rsreese:[EMAIL PROTECTED]:5060/rsreese context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) ;match_auth_username=yes ; if available, match user entry using the ; 'username' field from the authentication line ; instead of the From: field. allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) ; Default is enabled realm=ns1.neocipher.net ; Realm for digest authentication ; defaults to "asterisk". If you set a system name in ; asterisk.conf, it defaults to that system name ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port for unencrypted UDP ; and TCP sessions is 5060) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; You can specify port here too, like 123.123.123.123:5080 domain=neocipher.net [101] type=friend ; allows incoming and outgoing calls username=101 secret=pass mailbox=101 callerid=\"Stephen\" <101> host=dynamic nat=yes dtmfmode=rfc2833 canreinvite=no reinvite=no qualify=yes [102] type=friend ; allows incoming and outgoing calls username=102 secret=pass mailbox=102 callerid=\"Stephen\" <102> host=dynamic nat=yes dtmfmode=rfc2833 canreinvite=no reinvite=no qualify=yes [vitel-inbound] ;(exact format/casing required) type=friend host=inbound18.vitelity.net context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED]) username=rsreese secret=test allow=all ;insecure=very insecure = invite canreinvite=no [vitel-outbound] ;(exact format/casing required) type=friend host=outbound.vitelity.net context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED]) username=rsreese fromuser=rsreese trustrpid=yes sendrpid=yes secret=test allow=all canreinvite=no Here is the sip debug error: <------------> -- Executing [EMAIL PROTECTED]:1] Dial("SIP/102-08266f60", "'SIP/102',20") in new stack [Oct 19 16:21:21] WARNING[26690]: channel.c:3470 ast_request: No channel type registered for ''SIP' [Oct 19 16:21:21] WARNING[26690]: app_dial.c:1450 dial_exec_full: Unable to create channel of type ''SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup("SIP/102-08266f60", "") in new stack == Spawn extension (default, 102, 2) exited non-zero on 'SIP/102-08266f60' Scheduling destruction of SIP dialog 'NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.' in 32000 ms (Method: INVITE) ns1*CLI> <--- Reliably Transmitting (NAT) to 68.156.63.118:56558 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 68.156.63.118:56558;branch=z9hG4bK-d8754z-e0a3d830adb35401-1---d8754z-;received=68.156.63.118;rport=56558 From: <sip:[EMAIL PROTECTED]>;tag=7d39014c To: "102"<sip:[EMAIL PROTECTED]>;tag=as4f32f2a7 Call-ID: NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users